CodeBus
www.codebus.net
Search
Sign in
Sign up
Hot Search :
Source
embeded
web
remote control
p2p
game
More...
Location :
Home
Search - MATLAB c
Main Category
SourceCode
Documents
Books
WEB Code
Develop Tools
Other resource
Search - MATLAB c - List
[
Speech/Voice recognition/combine
]
libsvm-2.84-string
DL : 0
MATLAB版的SVM,使用相对于C来比较的方便,可以试一试
Date
: 2008-10-13
Size
: 28.03kb
User
:
xiaoxiao
[
Speech/Voice recognition/combine
]
fastica
DL : 0
FASTICA - Fast Independent Component Analysis % % FastICA for Matlab 7.x and 6.x % Version 2.5, October 19 2005 % Copyright (c) Hugo G鋠ert, Jarmo Hurri, Jaakko S鋜el� and Aapo Hyv鋜inen. % % FASTICA(mixedsig) estimates the independent components from given % multidimensional signals. Each row of matrix mixedsig is one % observed signal. FASTICA uses Hyvarinen s fixed-point algorithm,
Date
: 2008-10-13
Size
: 5.47kb
User
:
薛耀斌
[
Speech/Voice recognition/combine
]
HTK-3.0.tar
DL : 0
用于语音识别,基于HMM模型,用C++语言编写。可用连续语音识别-It is based on HMM Model and developed with C++ which could be used to continuous speech recognition.
Date
: 2025-12-27
Size
: 1.47mb
User
:
吴昊
[
Speech/Voice recognition/combine
]
hmm的c++语言实现
DL : 0
c++实现HMM,向前向后算法,Viterbi算法,Baum-Welch算法。其中包括用c++定义的HMM数据结构。全部是cpp和h的文件-c achieve HMM, forward backward algorithm, Viterbi algorithm, Baum-Welch algorithm. C including the use of the HMM definition data structure. Cpp all the documents and h
Date
: 2025-12-27
Size
: 8kb
User
:
宋敏
[
Speech/Voice recognition/combine
]
VQ方法做的说话人辨识vq
DL : 0
利用矢量量化(VQ)的方法实现的说话人辨认的程序,有兴趣者可以把它转换为C的-vector quantization (VQ) method to identify the Speaker of the proceedings, interested persons can transform it to C
Date
: 2025-12-27
Size
: 5kb
User
:
陈建文
[
Speech/Voice recognition/combine
]
HMM_CLUSTERING
DL : 0
HMM的C++详细源代码和矢量量化的C++源代码-HMM C source code and detailed vector quantization of C source code
Date
: 2025-12-27
Size
: 11kb
User
:
陆雯
[
Speech/Voice recognition/combine
]
FCM_HCM
DL : 0
模糊识别的一种分类方法,属于模糊C均值方法,里面包括了算法与具体步骤。欢迎下载-fuzzy identification of a classification method is Fuzzy C- Means, includes the algorithm and the concrete steps. Welcome to download
Date
: 2025-12-27
Size
: 1kb
User
:
王小然
[
Speech/Voice recognition/combine
]
dsds
DL : 0
基于改进LPCC和MFCC的汉语耳语音识别,在MATLAB上设计出了基于LPCC和MFCC的汉语语音孤立词识别/-LPCC and MFCC Based on Improved Chinese ear speech recognition, in the MATLAB design based on the LPCC and MFCC Chinese isolated word speech recognition /
Date
: 2025-12-27
Size
: 336kb
User
:
chenfeng
[
Speech/Voice recognition/combine
]
MMSE3in1
DL : 0
语音信号处理,MMSE降噪3合1,matlab源码,上传时间2008年5月26日-Speech Signal Processing, MMSE Noise Reduction 3-1, matlab source code, the upload time May 26, 2008
Date
: 2025-12-27
Size
: 10kb
User
:
意乱
[
Speech/Voice recognition/combine
]
mendelHMM
DL : 0
hmm的工具箱,里面有很全面的hmm的matlab程序包,适用于各种环境使用-hmm the toolbox, which has a very comprehensive hmm the matlab package, applicable to a variety of environmental use
Date
: 2025-12-27
Size
: 28kb
User
:
丁国梁
[
Speech/Voice recognition/combine
]
speech_processing(sola)
DL : 0
参考"基于SOLA的Pitch Scaling算法 范钰华(上海交通大学)",使用MATLAB实现SOLA算法,可以在变调不变速的基础下改变声调,程序比较易懂.适合初学者参考的,另外,VOICE_GUI是界面,介绍一些简单的语音处理,请大家多多指教.-Reference Based on the Pitch Scaling Algorithm SOLA Fan Yu-hua (Shanghai Jiaotong University) , the use of MATLAB to achieve SOLA algorithm sandhi can not change the basis of variable-speed voice, procedures relatively easy to understand. Suitable for beginners and reference, and the other, VOICE_GUI is the interface, some simple voice processing, please exhibitions.
Date
: 2025-12-27
Size
: 270kb
User
:
edwin wong
[
Speech/Voice recognition/combine
]
VAD
DL : 0
C语言实现的端点检测,效果很好的,和matlab效果很好的-C language to achieve the endpoint detection, the effect of good, and the effect of good matlab
Date
: 2025-12-27
Size
: 130kb
User
:
陈吉成
[
Speech/Voice recognition/combine
]
markdsp
DL : 0
语音降噪 滤波器 源代码 c语言-Voice noise reduction filter source code c language
Date
: 2025-12-27
Size
: 42kb
User
:
zsx
[
Speech/Voice recognition/combine
]
yinacf10
DL : 0
The Yin algorithm was developed by Alain de Cheveigné of IRCAM-CNRS and Hideki Kawahara of Wakayama University. It allows for real-time continuous (for each sample) fundamental frequency estimation. It features a very low error rate and few tuning parameters. This implementation is a C++ template, with EMM support for faster processing. Developped in Visual C++ 6.0, but should compile with other compilers.-The Yin algorithm was developed by Alain de Cheveigné of IRCAM-CNRS and Hideki Kawahara of Wakayama University. It allows for real-time continuous (for each sample) fundamental frequency estimation. It features a very low error rate and few tuning parameters. This implementation is a C++ template, with EMM support for faster processing. Developped in Visual C++ 6.0, but should compile with other compilers.
Date
: 2025-12-27
Size
: 12kb
User
:
_tika_
[
Speech/Voice recognition/combine
]
ss-improve
DL : 2
谱减法消除噪音的c代码,是在本站下载的源代码ss.tar.gz的基础上使用改进算法改进的,原程序只能消除最开始很小段的语音噪声。改进后的程序经测试,能够很好地消除语音噪声。在此感谢ss.tar.gz源程序的提供者。-C-code for spectrum-sub to cancel voice noise, can cancel noise very well.
Date
: 2025-12-27
Size
: 800kb
User
:
Guan
[
Speech/Voice recognition/combine
]
matsig-0[1][1].2.4
DL : 0
本程序用matlab设计了语音识别的程序,朋友要是需要c语言的请反编了-This program is designed to use matlab voice recognition program, friends, if needed c language, invites the Counter-compiled
Date
: 2025-12-27
Size
: 189kb
User
:
许愿
[
Speech/Voice recognition/combine
]
dianhuabohaoyuyinshibie
DL : 0
双音多频 DTMF( Dual Tone Multi-Frequency )信号,是用两个特定的单 音频率信号的组合来代表数字或功能。在 DTMF 电话机中有 16 个按键,其中 10 个数字键 0 — 9 , 6 个功能键 * 、 # 、 A 、 B 、 C 、 D 。其中 12 个按键是我们比较熟悉的按键,另外由第 4 列确定的按键作为保留,作为功能 键留为今后他用。 根据 CCITT 建议,国际上采用 697Hz 、 770Hz 、 852Hz 、 94lHz 低频群及 1209Hz 、 1336Hz 、 1477Hz 、 1633Hz 高频群。从低频群和高频群任意各抽出一种频率进行组合,共有 16 种组合,代表 16 种不同的数 字键或功能,每个按键唯一地由一组行频和列频组成-DTMF DTMF (Dual Tone Multi-Frequency) signal, is one of two specific A combination of audio frequency signals to represent the number or function. In the DTMF telephone has 16 buttons, which 10 number keys 0- 9, 6 function keys*,#, A, B, C, D. 12 of them Keys is more familiar with the keys, another set from the first four buttons as retention of a functional Key left for future uses. According to CCITT recommendations, the international use of 697Hz, 770Hz, 852Hz, 94lHz low frequency group and 1209Hz, 1336Hz, 1477Hz, 1633Hz frequency group. From the low group and high-frequency group out of any one frequency for each combination, a total of 16 combinations, the number of representatives of 16 different Character or function keys, each key uniquely by a group composed of row frequency and column frequency
Date
: 2025-12-27
Size
: 12kb
User
:
李小勇
[
Speech/Voice recognition/combine
]
CPPand-matlab
DL : 0
C++与matlab结合编程的语音识别实现-Combination of C++ programming with matlab implementation of speech recognition
Date
: 2025-12-27
Size
: 164kb
User
:
晴儿
[
Speech/Voice recognition/combine
]
twoSpk_unsupervised
DL : 0
无监督的两个说话人的语音分离程序。matlab/C混合编程,有说明文档,可直接使用,搞语音分离的可以下载看看。-Unsupervised speech separation for two speaker. Matlab/C mixed programming, there are documentation, can be used directly, and engage in voice separation can download to see.
Date
: 2025-12-27
Size
: 840kb
User
:
linlinchen
[
Speech/Voice recognition/combine
]
chenxu
DL : 0
(1)录制一段语音信号,完成对信号的采样,画出信号的时域波形和频谱图,确定信号的频谱范围; (2)给信号叠加噪声(噪声类型分为如下几种:a白噪声;b单频噪色(正弦干扰);c多频噪声(多正弦干扰);d其它干扰。),画出受噪声干扰的信号时域波形和频谱图; (3)采用窗函数法设计FIR低通滤波器,画出滤波器的频响特性图; (4)用所设计的滤波器对受噪声影响的信号进行滤波,画出滤波后语音信号的时域波形图和频谱图; (5)对滤波前后的信号进行对比,分析信号的变化;回放语音信号,并与原始语音信号对比。((1) record a speech signal, complete the sampling of the signal, draw the time domain waveform and frequency spectrum of the signal, and determine the spectrum range of the signal; (2) signal superposition noise (noise type is divided into the following: a white noise, B single frequency noise (sinusoidal interference); C multifrequency noise (multi sinusoidal interference); d other interference) Noise signal, time domain waveform and spectrogram are drawn; (3) the FIR low-pass filter is designed by using the window function method, and the frequency response characteristic of the filter is drawn; (4) the designed filter is used to filter the signal affected by the noise, and the time domain waveform and spectrogram of the filtered speech signal are drawn; (5) compare the signals before and after filtering, analyze the change of the signal, replay the speech signal, and compare with the original speech signal)
Date
: 2025-12-27
Size
: 13kb
User
:
华
«
1
2
»
CodeBus
is one of the largest source code repositories on the Internet!
Contact us :
1999-2046
CodeBus
All Rights Reserved.